1. Technical Field
The present invention relates generally to communication networks, and more particularly, to a CMDA2000 network that uses Session Initiation Protocol (SIP) for call control over a wireless communication network.
2. Related Art
Traditional telephone networks, including the Public Switched Telephone Network (PSTN) and Signaling System Number 7 (SS7) networks, have provided closed systems that enabled users to achieve added capabilities beyond merely connecting a call. Initially, message storage capabilities such as those provided by answering machines and voice mail services (in SS7 networks) were popular. Since then, many other services have been developed as SS7 and other intelligent networks (IN and AIN) have gained widespread popularity. With the advent of Internet telephony (the use of an Internet, public or private, to transport voice services), a need to provide voice services, beyond those thought of as traditional services, has been realized.
During the past few years, Internet telephony has evolved from being a novelty for the technically-oriented seeking party conversation material to a technology that, in the not too distant future, may largely replace the existing PSTN. Supporting the widespread use of Internet telephony requires a host of standardized protocols to ensure that audio and video data are routed correctly and are transported with a specified quality of service (QoS). Call control protocols are of particular interest because they are the basis that allow for advanced services such as number portability, multiparty conferencing, voice/video mail, and automatic call distribution.
Session Initiation Protocol (SIP) is an application-layer control (signaling) protocol (defined in RFC3261) designed for creating, modifying, and terminating sessions (e.g., Internet telephony) with one or more participants. Its strengths include simplicity, scalability, extensibility, and modularity. As a result, increasing interest in SIP is being realized as the SIP standards and protocol requirements develop into maturity.
As a traditional text-based Internet protocol, SIP resembles the hypertext transfer protocol (HTTP) and simple mail transfer protocol (SMTP). SIP uses Session Description Protocol (SDP) for describing multimedia sessions for the purposes of session announcement, session invitation, and other forms of multimedia session initiation. SIP, however, is independent of the transport layer (i.e., IP) and also supports mechanisms for encryption and authentication.
Current CDMA related standards for circuit-switched networks only address using network protocols such as SS7 for setting up a voice service. CDMA circuit-switched networks that use network protocols, such as SS7, for call control have several drawbacks for mobile-to-mobile calls. One that is significant is that the voice bearer path (path actually carrying the voice between the two mobiles) for mobile-to-mobile calls will require two transcodings by two transcoders. A transcoder is a device that changes data (e.g., audio) from one format to another (e.g., from EVRC to G.711). For audio data streams transcoding is undesirable for it decreases voice quality and increases the time delay between the end users. A need exists, therefore, for a system and method that minimizes the number of transcoders between a calling party and a called party.
A shortcoming of LMSD systems is ringback (or call progress tones) to the calling party. Ringback is traditionally generated by the network supporting the called party. As compared to circuit-switched networks, the timing of when ringback is sent by the network supporting the called party back to the calling party is somewhat longer in time and can result in discomfort to the calling party (i.e., the feeling on the part of the user that nothing is happening). This increase in time is a result of the method of establishing the voice-bearer path. The voice bearer path, using SIP/SIP-T as the call control protocol, cannot be established until an exchange of bearer related information between the called party control entity and the calling party control entity occurs. A need exists, therefore to provide ringback to the calling party in an improved manner.